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37472 discussions

continue processing route script if no response from gateway
by JR Richardson 28 May '10

28 May '10

Hi All, I am lab testing carrierroute modue on kamailio 1.5.4-notls (i386/linux) and have a question on how to continue processing a call if kamailio sends a call with t_relay() but does not get a response from the gateway. I read about the timers in the tm module and set them pretty low in the config file: # ----- tm params ----- modparam("tm", "fr_timer", 2) modparam("tm", "fr_inv_timer_next", 3) But nothing happens in the routing script after an invite is sent and there is no response. Debug just shows the invite to the gateway being re-transmited several times. Any ideas? Thanks. JR -- JR Richardson Engineering for the Masses

3 7

kamailio 3.0.2 for Thu, May 27, 2010
by Daniel-Constantin Mierla 28 May '10

28 May '10

Hello, I am planning to release kamailio 3.0.2 this Thursday. There were some fixes since 3.0.1 that worth to be packaged. If you have major reports for current stable version, please write to sr-dev(a)lists.sip-router.org Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Miami, Fl, USA - June 21-23, 2010 http://www.asipto.com/index.php/kamailio-advanced-training/

3 7

RPID Question
by dotnetdub 28 May '10

28 May '10

Hi All, Kamailio 3 I currently store a default RPID in our database. We allow some asterisk customers to send us the RPID. I want to store the RPID sent by the customer on the radius record. We test: if!(is_present_hf("Remote-Party-ID")) { #RPID Not Set by Asterisk - Will set now from DB append_rpid_hf(); } Do I use textops to get the RPID that the Asterisk sends and then overwrite what we read from the database? $avp(s:rpid) = ? Not really sure how to extract the RPID from the SIP message sent from the asterisk. TIA. Stephen.

3 2

new feature: embedded http 1.0 server
by Daniel-Constantin Mierla 28 May '10

28 May '10

Hello, there is a new module named xhttp which provides an embedded http server inside the SIP server. It reuses the transport layer from the core, so you can send http requests via TCP and TLS. The port is the same as for SIP traffic. The documentation of the module is available at: http://sip-router.org/docbook/sip-router/branch/master/modules/xhttp/xhttp.… Since sip and http are very similar, you can use other modules in handling the http requests. For example, you can do www-authentication: event_route[xhttp:request] { if (!www_authorize("xcap", "subscriber")) { www_challenge("sip-router.org", "0"); exit; } xhttp_reply("200", "OK", "text/html", "<html><body>Welcome $au, you are connected from $si:$sp</body></html>"); } allowing in this way only local SIP users to do http requests. For now supports only http 1.0, meaning it cannot take chunked requests (usually happening in PUT requests with large body). Hope it is useful for some of you. Its main goal right now is to offer the transport layer for embedded xcap server (the new xcap_server module), but it can be used for service monitoring, server control, etc. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Miami, Fl, USA - June 21-23, 2010 http://www.asipto.com/index.php/kamailio-advanced-training/

1 0

Kamailio v3.0.2 Released
by Daniel-Constantin Mierla 27 May '10

27 May '10

Hello, Kamailio v3.0.2 was just released. It is a patch release that includes the fixes done since release of v3.0.1. You should consider to upgrade if you run v3.0.0 or or v3.0.2, there is no change that you should do to configuration file or database, just reinstall the binaries. Source tarballs are available at: http://www.kamailio.org/pub/kamailio/3.0.2/src/ Detailed changelog: http://www.kamailio.org/pub/kamailio/3.0.2/ChangeLog Download via GIT: # git clone –depth 1 git://git.sip-router.org/sip-router kamailio # cd kamailio # git checkout -b kamailio_3.0 origin/kamailio_3.0 Binaries and packages will be uploaded at : http://www.kamailio.org/pub/kamailio/3.0.2/ Modules’ documentation: http://www.kamailio.org/docs/modules/3.0.x/ What is new in 3.0.x release series is summarized in the announcement of v3.0.0: http://www.kamailio.org/w/kamailio-openser-v3.0.0-release-notes/ If you are eager to see what's up to next major release (3.1.0), visit: http://sip-router.org/wiki/features/new-in-devel Enjoy, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

1 0

Carrierroute failover domain not working
by JR Richardson 27 May '10

27 May '10

Hi All, I'm testing carrierroute modue in kamailio 1.5.4-notls (i386/linux). I have things working like this: sipp><kamailio carrierroute>< 2 asterisk servers A & B even distribution of calls So A & B are in carrier1 domain1, I have failureroute setup with 2 more asterisk servers C& D in carrier1 domain2. While I'm processing calls from sipp and sending to A & B, I can fail A and then all calls go to B as expected. But if I fail both A & B, the calls fail, so I'm guessing I don't have the faliure_route configured correctly. Here is my current config: route { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if ($rU==NULL) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; if (!allow_trusted()) { sl_send_reply("403", "Forbidden"); exit; }; if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); route(2); } route[1] { if (!t_relay()) { sl_reply_error(); } exit; } route[2] { if(!cr_route("1", "1", "$rU", "$rU", "call_id")){ sl_send_reply("403", "Not allowed"); } else { t_on_failure("2"); route(1); } } failure_route[2] { if (t_check_status("408|5[0-9][0-9]")) { if(!cr_route("1", "2", "$rU", "$rU", "call_id")){ t_reply("403", "Not allowed"); } else { t_relay(); } } } I think the database entries are setup ok, I used the example in the module doc, but I can share that if needed. Any guidance will be appriciated. Thanks. JR -- JR Richardson Engineering for the Masses

2 4

functions for modifying pAsserted
by Alex rsm 27 May '10

27 May '10

Hi, I couldn't find the module that has the functions for modifying pAsserted field. Your help would be appreciated. Thanks, AR _________________________________________________________________ Hotmail has tools for the New Busy. Search, chat and e-mail from your inbox. http://www.windowslive.com/campaign/thenewbusy?ocid=PID28326::T:WLMTAGL:ON:…

3 2

Siremis 1.0 login fails (or, does nothing)
by Joe Micciche 27 May '10

27 May '10

RHEL5.5 server Kamailio 3.0 built from source via Git Apache/2.2.3 PHP 5.3.2 MySQL 5.1.47 chown -R apache /var/www/html/siremis-1.0.0, apache is the httpd user. Used the instructions from Asipto's site for Siremis install and configure, added the alias to httpd.conf, Config.xml is: <?xml version="1.0" standalone="no"?> <Config> <DataSource> <Database Name="Default" Driver="Pdo_Mysql" Server="127.0.0.1" DBName="openser" User="openser" Password="*****" /> <Database Name="Siremis" Driver="Pdo_Mysql" Server="127.0.0.1" DBName="openser" User="openser" Password="*****" /> </DataSource> </Config> I navigate to the Siremis page, http://10.5.10.21/siremis/ and get the login screen, enter the login credentials, Submit, and nothing happens. httpd/error_log: [Wed May 26 12:49:45 2010] [error] [client 10.3.226.168] File does not exist: /var/www/html/siremis-1.0.0/siremis-web/js/sound.js, referer: http://10.5.10.21/siremis/bin/ [Wed May 26 12:49:45 2010] [error] [client 10.3.226.168] File does not exist: /var/www/html/siremis-1.0.0/siremis-web/js/sound.js, referer: http://10.5.10.21/siremis/bin/ I have wiped Siremis and done a fresh install, restarted httpd, and still nothing. Does anyone have ideas or suggestions on where else to look? -- ================================================================== Joe Micciche jmiccich(a)redhat.com Red Hat, Inc. http://www.redhat.com Senior Communications Engineer X(81) 44554 +1.919.754.4554 ==================================================================

6 15

t_check_trans return false on CANCEL
by Ernest Mavrel 27 May '10

27 May '10

3 2

MAX_BRANCHES
by Alex Balashov 27 May '10

27 May '10

This may seem like an absurd question, but is there a particular reason why MAX_BRANCHES is #defined as 12 in config.h? Any particular reason for 12? Is it okay to make it higher? -- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

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